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CallManager Express 7.1 with Unity Express forward problem IOS 15.0 2900 router

dinorubio
Level 1
Level 1

Hi there,

I'm trying to get calls forwarded to voice mail ISM0/0 and calls are dropped.

Normally this ip phones receive calls with g729a from H.323 signaling, i configured a forward no answer to voice mail. When the call is not answered it tries to forward to voice mail with SIP signaling, but the call fails (i believe that transcoding is not coming up with the interworking). I have interworking configured and transcoding - i will post the config.

If i put an incoming dial-peer (dial-peer 7 voip) with g711ulaw to that E.164 phone number (to get calls forwarded to voice mail but with the codec g711ulaw from the PSTN) calls enter voice mail with sucess. But if i take out that dial-peer that call fails, as i previously described.

P.S. - i configured two type of transcoding for testing and both do normal transcoding.. as you can see below

#sh sccp connections

sess_id    conn_id      stype mode     codec   sport rport ripaddr

7          8            xcode sendrecv g729    32636 2000  134.134.134.88
7          7            xcode sendrecv g711u   16386 2000  134.134.134.88

Total number of active session(s) 1, and connection(s) 2

Thanks everybody!

!

voice call carriercapacity active

voice rtp send-recv

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

!

sccp local Loopback0

sccp ccm 134.134.134.88 identifier 2 version 7.0

sccp ip precedence 4

sccp

!

sccp ccm group 2

bind interface Loopback0

associate ccm 2 priority 1

associate profile 23 register CCME_TRX2

associate profile 21 register CCME_TRX

associate profile 22 register CCME_CFB

keepalive retries 2

switchover method immediate

switchback method graceful

switchback interval 5

signaling dscp af41

!

dspfarm profile 21 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

maximum sessions 3

associate application SCCP

shutdown

!

dspfarm profile 23 transcode universal

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

maximum sessions 1

associate application SCCP

!

dspfarm profile 22 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 1

associate application SCCP

shutdown

!

dial-peer voice 1 voip

description Outgoing Public Gateway                         

translation-profile outgoing to_pstn

destination-pattern 0T

session target ras

voice-class codec 1

dtmf-relay h245-signal h245-alphanumeric

ip qos dscp af41 signaling

no vad

!

dial-peer voice 2 voip

description TO_CCME_PHONES

translation-profile incoming inbound

incoming called-number 21004282.

voice-class codec 1

dtmf-relay h245-signal h245-alphanumeric

ip qos dscp af41 signaling

no vad

!

dial-peer voice 5 voip

destination-pattern 1...

session protocol sipv2

session target ipv4:134.134.134.85

dtmf-relay sip-notify

codec g711ulaw

no vad

!

dial-peer voice 6 voip

description SIP dial peer to Cisco Unity Express

translation-profile incoming inbound

destination-pattern 21004282.

session protocol sipv2

session target ipv4:134.134.134.85

dtmf-relay sip-notify

codec g711ulaw

no vad

!

dial-peer voice 7 voip

description VOICE MAIL INCOMING

translation-profile incoming inbound

shutdown

no modem passthrough

session protocol sipv2

incoming called-number 210042822

codec g711ulaw

no vad 

!

telephony-service

sdspfarm conference mute-on 123 mute-off 321

sdspfarm units 3

sdspfarm transcode sessions 4

sdspfarm tag 1 CCME_CFB

sdspfarm tag 2 CCME_TRX

sdspfarm tag 3 CCME_TRX2

conference hardware

max-ephones 10

max-dn 100

ip source-address 134.134.134.88 port 2000 strict-match

calling-number local

timeouts interdigit 4

system message CCME 7.1 IOS 15.0

url services [http://134.134.134.85/voiceview/common/login.do | http://134.134.134.85/voiceview/common/login.do]

url authentication [http://134.134.134.88/CCMCIP/authenticate.asp | http://134.134.134.88/CCMCIP/authenticate.asp]

cnf-file location flash:

cnf-file perphone

user-locale PT

network-locale PT

load 7921 CP7921G-1.2.1.LOADS

load 7941 SCCP41.8-4-2S

time-format 24

date-format dd-mm-yy

voicemail 1020

max-conferences 4 gain -6

call-forward pattern 0.........

call-forward pattern ....

moh music-on-hold.au

dn-webedit

time-webedit

transfer-system full-consult

transfer-pattern 0.........

transfer-pattern ....

secondary-dialtone 0

create cnf-files version-stamp Jan 01 2002 00:00:00

!

6 Replies 6

acampbell
VIP Alumni
VIP Alumni

Hi,

From your config I can see:-

!

telephony-service

voicemail 1020

!

!

That means you should have a SIP dial peer pointing to the VMAIL

!
dial-peer voice 1020 voip
destination-pattern 1020
voice-class codec 1
session protocol sipv2
session target ipv4:YOUR VMAIL IP ADD

dtmf-relay sip-notify
no vad
!

HTH

Alex

Regards, Alex. Please rate useful posts.

Actually, he has that covered using dial-peer voice 5 voip.

Dial-peer 6 though sure has me scratching my head.  Not sure what that one is for. 

Try this:

Get rid of this DSPFarm profile all together:

dspfarm profile 21 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

maximum sessions 3

associate application SCCP

shutdown

!

Modify this one as follows:

dspfarm profile 23 transcode universal

codec g711ulaw

codec g711alaw

codec g729r8   <=   all other codecs removed

maximum sessions 4

associate application SCCP

I will say that normally you don't need to do this.  I'm not sure why you are needing to.  You should be able to get this to work without having any transcode resources configured PROVIDED all the calls are coming in from the PSTN, and not coming in as G729 over the WAN.

Cliff

Please rate helpful posts

Hi there,

The calls are coming from PSTN with g729 codec and failing. Only when i activate dial-peer 7 the calls are successful because all legs are g711ulaw and no transcoding is needed.

Dino

Have you done a debug on the dial-peers to determine which dial-peers are getting engaged during setup?  I'm wondering if you're hitting the default dial peer, which will NOT negotiate codec.

Hi there, the call is coming in through dial-peer 2,

   Calling Number=300504602,(Calling Name=)(TON=National, NPI=ISDN, Screening=User, Passed, Presentation=Allowed),

   Called Number=210042822(TON=National, NPI=ISDN),

   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,

   Incoming Dial-peer=2, Progress Indication=NULL(0), Calling IE Present=TRUE,

Dino

Hi there,

Anyone has any ideas?

Thanks,

Dino

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